Asterisk don't route calls to the destination
Posted: Mon Dec 09, 2019 7:04 am
Hello
I have installed Vicidial. I launched a call from (phone 101) to (phone 100) to test the regidered phones, but i get this error in the Asterisk logs (in red). Any help will be appreciated.
localhost*CLI>
== Using SIP RTP CoS mark 5
> 0x7fc800070c30 -- Strict RTP learning after remote address set to: 172.17.2.120:20000
-- Executing [100@default:1] Dial("SIP/101-00000004", "SIP/100|60|") in new stack
== Using SIP RTP CoS mark 5
[Dec 9 12:49:40] ERROR[13135][C-00000004]: netsock2.c:305 ast_sockaddr_resolve: getaddrinfo("100|60|", "(null)", ...): Name or service not known
[Dec 9 12:49:40] WARNING[13135][C-00000004]: chan_sip.c:6316 create_addr: No such host: 100|60|
[Dec 9 12:49:40] WARNING[13135][C-00000004]: app_dial.c:2527 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [100@default:2] Goto("SIP/101-00000004", "default,85026666666666100,1") in new stack
-- Goto (default,85026666666666100,1)
-- Executing [85026666666666100@default:1] Wait("SIP/101-00000004", "1") in new stack
-- Executing [85026666666666100@default:2] VoiceMail("SIP/101-00000004", "100,u") in new stack
> 0x7fc800070c30 -- Strict RTP switching to RTP target address 172.17.2.120:20000 as source
-- <SIP/101-00000004> Playing 'vm-theperson.gsm' (language 'en')
== Spawn extension (default, 85026666666666100, 2) exited non-zero on 'SIP/101-00000004'
[Dec 9 12:49:43] WARNING[13135][C-00000004]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
-- Executing [h@default:1] AGI("SIP/101-00000004", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL---------------)") in new stack
-- <SIP/101-00000004>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ----------) completed, returning 0
localhost*CLI>
I have installed Vicidial. I launched a call from (phone 101) to (phone 100) to test the regidered phones, but i get this error in the Asterisk logs (in red). Any help will be appreciated.
localhost*CLI>
== Using SIP RTP CoS mark 5
> 0x7fc800070c30 -- Strict RTP learning after remote address set to: 172.17.2.120:20000
-- Executing [100@default:1] Dial("SIP/101-00000004", "SIP/100|60|") in new stack
== Using SIP RTP CoS mark 5
[Dec 9 12:49:40] ERROR[13135][C-00000004]: netsock2.c:305 ast_sockaddr_resolve: getaddrinfo("100|60|", "(null)", ...): Name or service not known
[Dec 9 12:49:40] WARNING[13135][C-00000004]: chan_sip.c:6316 create_addr: No such host: 100|60|
[Dec 9 12:49:40] WARNING[13135][C-00000004]: app_dial.c:2527 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [100@default:2] Goto("SIP/101-00000004", "default,85026666666666100,1") in new stack
-- Goto (default,85026666666666100,1)
-- Executing [85026666666666100@default:1] Wait("SIP/101-00000004", "1") in new stack
-- Executing [85026666666666100@default:2] VoiceMail("SIP/101-00000004", "100,u") in new stack
> 0x7fc800070c30 -- Strict RTP switching to RTP target address 172.17.2.120:20000 as source
-- <SIP/101-00000004> Playing 'vm-theperson.gsm' (language 'en')
== Spawn extension (default, 85026666666666100, 2) exited non-zero on 'SIP/101-00000004'
[Dec 9 12:49:43] WARNING[13135][C-00000004]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
-- Executing [h@default:1] AGI("SIP/101-00000004", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL---------------)") in new stack
-- <SIP/101-00000004>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ----------) completed, returning 0
localhost*CLI>